asterisk/asterisk.git
23 months agoalembic/res_pjsip: Add "webrtc" configuration option
Kevin Harwell [Wed, 2 Aug 2017 14:43:56 +0000 (09:43 -0500)]
alembic/res_pjsip: Add "webrtc" configuration option

When the "webrtc" option was added in res_pjsip it was not added to the alembic
scripts. This patch adds the option for alembic.

Also, changed the sorcery configuration type to an OPT_YESNO_T value instead of
an OPT_BOOL_T so if this field is ever written to a database it will write out
the correct value.

ASTERISK-27119 #close

Change-Id: I3e199f060aea25e193c439fc5cf96be4d3ed1c7b

23 months agores_pjsip_pidf_eyebeam_body_supplement: Correct status presentation
Sean Bright [Wed, 26 Jul 2017 13:48:29 +0000 (09:48 -0400)]
res_pjsip_pidf_eyebeam_body_supplement: Correct status presentation

This change fixes PIDF content generation when the underlying device
state is considered in use. Previously it was incorrectly marked
as closed meaning they were offline/unavailable. The code now
correctly marks them as open.

Additionally:

  * Generate an XML element for our activity instead of a using a text
    node.

  * Consider every extension state other than "unavailable" to be 'open'
    status.

  * Update the XML namespaces and structure to reflect those
    documented in RFC 4480

  * Use 'on-the-phone' (defined in RFC 4880) instead of 'busy' as the
    "in use" activity. This change results in eyeBeam using the
    appropriate icon for the watched user.

This was tested on eyeBeam 1.5.20.2 build 59030 on Windows.

ASTERISK-26659 #close
Reported by: Abraham Liebsch
patches:
  ASTERISK-26659.diff submitted by snuffy (license 5024)

Change-Id: I6e5ad450f91106029fb30517b8c0ea0c2058c810

23 months agores_pjsip: Add support for dnsmgr to external_media_address.
Joshua Colp [Sun, 23 Jul 2017 23:34:32 +0000 (23:34 +0000)]
res_pjsip: Add support for dnsmgr to external_media_address.

The "external_media_address" option on transports is now
resolved using dnsmgr. This allows it to be automatically
refreshed regularly if refreshes are enabled in dnsmgr.
If the system is using a dynamic IP address a dynamic DNS
hostname can be provided to keep the IP address up to
date.

Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2

23 months agoFix compiler warnings on Fedora 26 / GCC 7.
Corey Farrell [Fri, 28 Jul 2017 01:58:22 +0000 (21:58 -0400)]
Fix compiler warnings on Fedora 26 / GCC 7.

GCC 7 has added capability to produce warnings, this fixes most of those
warnings.  The specific warnings are disabled in a few places:

* app_voicemail.c: truncation of paths more than 4096 chars in many places.
* chan_mgcp.c: callid truncated to 80 chars.
* cdr.c: two userfields are combined to cdr copy, fix would break ABI.
* tcptls.c: ignore use of deprecated method SSLv3_client_method().

ASTERISK-27156 #close

Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88

23 months agoapp_queue: Add announce-position-only-up option
Sean Bright [Wed, 26 Jul 2017 14:27:00 +0000 (10:27 -0400)]
app_queue: Add announce-position-only-up option

Setting this option will cause the Queue application to only announce
the caller's position if it has improved since the last time that we
announced it.

Change-Id: I173a124121422209485b043e2bf784f54242fce6

23 months agobundled_pjproject: Improve SSL/TLS error handling
George Joseph [Thu, 27 Jul 2017 11:35:51 +0000 (05:35 -0600)]
bundled_pjproject:  Improve SSL/TLS error handling

OpenSSL has 2 levels or error processing.  It's possible for the
top layer to return SSL_ERROR_SYSCALL but the lower layer return
no error, in which case processing should continue.  Only the top
layer was being examined though so connections were being torn
down when they didn't need to be.  This patch adds the examination
of the lower level codes, and if they return no errors, allows
processing to continue.

ASTERISK-27001
Reported-by: Ian Gilmour
patches:
pjproject-2.6.patch submitted by Ian Gilmour (license 6889)

Updated-by: George Joseph and Sauw Ming (Teluu)

Merged to upstream pjproject on 7/27/2017 (commit 5631)

Change-Id: I23844ca0c68ef1ee550f14d46f6dae57d33b7bd2

23 months agochan_pjsip: add a new function PJSIP_DTMF_MODE
Torrey Searle [Mon, 26 Jun 2017 12:52:52 +0000 (14:52 +0200)]
chan_pjsip: add a new function PJSIP_DTMF_MODE

This function is a replica of SIPDtmfMode, allowing the DTMF mode of a
PJSIP call to be modified on a per-call basis

ASTERISK-27085 #close

Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612

23 months agores_rtp_asterisk: Fix mapping of pjsip's ICE roles to ours
Sean Bright [Tue, 25 Jul 2017 20:17:45 +0000 (16:17 -0400)]
res_rtp_asterisk: Fix mapping of pjsip's ICE roles to ours

Change-Id: Ia578ede1a55b21014581793992a429441903278b

23 months agoMerge "Core: Add support for systemd socket activation."
Jenkins2 [Wed, 26 Jul 2017 14:17:40 +0000 (09:17 -0500)]
Merge "Core: Add support for systemd socket activation."

23 months agoMerge "bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues."
Joshua Colp [Wed, 26 Jul 2017 13:31:13 +0000 (08:31 -0500)]
Merge "bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues."

23 months agoMerge "res_stasis_device_state: Unsubscribe should remove old subscriptions"
Joshua Colp [Wed, 26 Jul 2017 13:27:31 +0000 (08:27 -0500)]
Merge "res_stasis_device_state: Unsubscribe should remove old subscriptions"

23 months agoMerge "SDP: Create declined m= SDP lines using remote SDP if applicable."
Joshua Colp [Wed, 26 Jul 2017 13:20:38 +0000 (08:20 -0500)]
Merge "SDP: Create declined m= SDP lines using remote SDP if applicable."

23 months agoMerge "SDP: Rework SDP offer/answer model and update capabilities merges."
Joshua Colp [Wed, 26 Jul 2017 13:20:35 +0000 (08:20 -0500)]
Merge "SDP: Rework SDP offer/answer model and update capabilities merges."

23 months agoMerge "app_voicemail.c: Allow mailbox entry on authentication retry prompt."
Jenkins2 [Wed, 26 Jul 2017 11:49:41 +0000 (06:49 -0500)]
Merge "app_voicemail.c: Allow mailbox entry on authentication retry prompt."

23 months agoMerge "core: Add VP9 passthrough support."
Jenkins2 [Tue, 25 Jul 2017 15:37:45 +0000 (10:37 -0500)]
Merge "core: Add VP9 passthrough support."

23 months agores_stasis_device_state: Unsubscribe should remove old subscriptions
Sergej Kasumovic [Thu, 20 Jul 2017 13:08:05 +0000 (15:08 +0200)]
res_stasis_device_state: Unsubscribe should remove old subscriptions

Case scenario with Applications ARI:

* Once you subscribe to deviceState with Applications REST API, it will be
added into subscription pool.

* When you unsubscribe it will remove from the device_state_subscription
hash table but not from the subscription pool.

* When you subscribe again, it will add it to pool again.

* Now you will have two subscriptions and you will receive same event
twice.

This fix should now remove deviceState subscription from pool and it
should fix unsubscribe on deviceState.

ASTERISK-27130 #close

Change-Id: I718b70d770a086e39b4ddba4f69a3c616d4476c4

23 months agoMerge "say.c: Fix file locations for second, seconds, minute, minutes files"
George Joseph [Tue, 25 Jul 2017 12:45:00 +0000 (07:45 -0500)]
Merge "say.c: Fix file locations for second, seconds, minute, minutes files"

23 months agocore: Add VP9 passthrough support.
Joshua Colp [Mon, 24 Jul 2017 18:30:59 +0000 (18:30 +0000)]
core: Add VP9 passthrough support.

This change adds VP9 as a known codec and creates a cached
"vp9" media format for use.

Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc

23 months agoMerge "format.h: Fix a few minor errors in comments."
Jenkins2 [Mon, 24 Jul 2017 15:54:30 +0000 (10:54 -0500)]
Merge "format.h: Fix a few minor errors in comments."

23 months agoMerge "Update make_ari_stubs in master to make the version 16"
Joshua Colp [Mon, 24 Jul 2017 12:41:43 +0000 (07:41 -0500)]
Merge "Update make_ari_stubs in master to make the version 16"

23 months agoMerge "Restore the incorrectly deleted spandspflow2pcap.log"
Jenkins2 [Mon, 24 Jul 2017 12:05:59 +0000 (07:05 -0500)]
Merge "Restore the incorrectly deleted spandspflow2pcap.log"

23 months agoapp_voicemail.c: Allow mailbox entry on authentication retry prompt.
Richard Mudgett [Wed, 19 Jul 2017 23:11:19 +0000 (18:11 -0500)]
app_voicemail.c: Allow mailbox entry on authentication retry prompt.

The following testsuite voicemail tests were failing to re-enter the
mailbox after the first login attempt.

tests/apps/voicemail/authenticate_invalid_mailbox
tests/apps/voicemail/authenticate_invalid_password

The tests were noting the start of the vm-incorrect-mailbox prompt and
immediately sending the mailbox for the next login attempt.  Since the
invalid message playback had to complete before the digits were
recognized, the test passed for the wrong reason and added approximately
20 seconds to the test times.

* Allow the vm-incorrect-mailbox prompt to get interrupted by the mailbox
digits like the initial vm-login prompt so the tests are able to enter the
intended mailbox.

Change-Id: I1dc53fe917bfe03a4587b2c4cd24c94696a69df8

23 months agoformat.h: Fix a few minor errors in comments.
Matthew Fredrickson [Fri, 21 Jul 2017 20:57:46 +0000 (15:57 -0500)]
format.h: Fix a few minor errors in comments.

A few minor problems were found in comments in format.h.  This patch fixes them.

Change-Id: I07f0bdb47b93359b361c4c3d8ecc87cd3199dd94

23 months agosay.c: Fix file locations for second, seconds, minute, minutes files
Rusty Newton [Fri, 14 Jul 2017 18:47:50 +0000 (13:47 -0500)]
say.c: Fix file locations for second, seconds, minute, minutes files

The seconds and minutes files have always existed in the base language
directory of the Core package. So say.c has always been calling the wrong
location (under digits/) for those two files and in the case of second and
minute they didn't exist in the Core packages at all.

The 1.6 sounds release moves the second and minute files into Core from
Extra for the languages that already had them. A future release will include
the second and minute files for languages that didn't already have them.

This patch just changes all the target locations for second, seconds,
minute, and minutes that were under the digits subdir to be under the root of
sounds instead. Which is where the sounds will be for some languages after 1.6
sounds and for all languages after a future release.

ASTERISK-25810 #close

Change-Id: I05d9d4bee6a7237030530a46e7eb3df15f13f702
Reported-by: Nicolas Riendeau

23 months agoSounds: Update Makefile for Extra sounds 1.5.1 release
Rusty Newton [Fri, 21 Jul 2017 19:20:10 +0000 (14:20 -0500)]
Sounds: Update Makefile for Extra sounds 1.5.1 release

Incrementing version for the Extra sounds release. 1.5.1 Extra sounds
removes two prompts that were moved into the Core packages in the 1.6 Core
sounds release.

ASTERISK-27142 #close

Change-Id: I82f017812b0ea9599e19dd4635afd55611f13ee7

23 months agoUpdate make_ari_stubs in master to make the version 16
George Joseph [Fri, 21 Jul 2017 16:17:38 +0000 (10:17 -0600)]
Update make_ari_stubs in master to make the version 16

Ready for next major version

Change-Id: If9dc99b3b78768529e69a297d8f87e23582ca6d0

23 months agoRestore the incorrectly deleted spandspflow2pcap.log
George Joseph [Fri, 21 Jul 2017 16:24:24 +0000 (10:24 -0600)]
Restore the incorrectly deleted spandspflow2pcap.log

Change-Id: Iafe78cf0fb1e7064223d4dea279eeb776c8fa8e5

23 months agoMerge "corosync: Fix corosync library name in configure.ac"
George Joseph [Fri, 21 Jul 2017 11:54:00 +0000 (06:54 -0500)]
Merge "corosync: Fix corosync library name in configure.ac"

23 months agoMerge "Update AMI and ARI versions for master/15 and update UPDATE.txt"
Jenkins2 [Thu, 20 Jul 2017 17:17:48 +0000 (12:17 -0500)]
Merge "Update AMI and ARI versions for master/15 and update UPDATE.txt"

23 months agoMerge "pjsip: Increase maximum packet size."
George Joseph [Thu, 20 Jul 2017 16:08:44 +0000 (11:08 -0500)]
Merge "pjsip: Increase maximum packet size."

23 months agoUpdate AMI and ARI versions for master/15 and update UPDATE.txt
George Joseph [Thu, 20 Jul 2017 15:52:38 +0000 (09:52 -0600)]
Update AMI and ARI versions for master/15 and update UPDATE.txt

AMI goes from 3.2.0 to 4.0.0
ARI goes from 2.0.0 to 3.0.0

Copied UPGRADE.txt -> UPGRADE-15.txt
Created new UPGRADE.txt
Removed a log file that was accidentally checked in a while ago

Change-Id: I1c794f910038459b13e16f9c3a12c44e56f142f7

23 months agocorosync: Fix corosync library name in configure.ac
Sean Bright [Thu, 20 Jul 2017 14:57:08 +0000 (10:57 -0400)]
corosync: Fix corosync library name in configure.ac

Also add new corosync packages to install_prereq.

Reported by Travis Ryan in #asterisk-dev

Change-Id: Ib861c95ba630fed62dc54e56784ad8446ed9d2db

23 months agoMerge "core: Add digit filtering to ast_waitfordigit_full"
Joshua Colp [Wed, 19 Jul 2017 18:09:56 +0000 (13:09 -0500)]
Merge "core: Add digit filtering to ast_waitfordigit_full"

23 months agoMerge "app_playback.c: Use the timezonename parameter"
George Joseph [Wed, 19 Jul 2017 17:11:09 +0000 (12:11 -0500)]
Merge "app_playback.c: Use the timezonename parameter"

23 months agoMerge "bridge_softmix: Use removed stream spots when renegotiating."
Jenkins2 [Wed, 19 Jul 2017 15:42:51 +0000 (10:42 -0500)]
Merge "bridge_softmix: Use removed stream spots when renegotiating."

23 months agoMerge "core: Add PARSE_TIMELEN support to ast_parse_arg and ACO."
Jenkins2 [Wed, 19 Jul 2017 14:25:59 +0000 (09:25 -0500)]
Merge "core: Add PARSE_TIMELEN support to ast_parse_arg and ACO."

23 months agobridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
Joshua Colp [Mon, 17 Jul 2017 16:01:24 +0000 (16:01 +0000)]
bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.

This change does a few things to improve packet loss and renegotiation:

1. On outgoing RTP streams we will now properly reflect out of order
packets and packet loss in the sequence number. This allows the
remote jitterbuffer to better reorder things.

2. Video updates can now be discarded for a period of time
after one has been sent to prevent flooding of clients.

3. For declined and removed streams we will now release any
media session resources associated with them. This was not
previously done and caused an issue where old state was being
used for a new stream.

4. RTP bundling was not actually removing bundled RTP instances
from the parent. This has been resolved by removing based on
the RTP instance itself and not the SSRC.

5. The code did not properly handle explicitly unbundling an
RTP instance from its parent. This now works as expected.

ASTERISK-27143

Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45

23 months agopjsip: Increase maximum packet size.
Benjamin Keith Ford [Tue, 18 Jul 2017 20:04:44 +0000 (15:04 -0500)]
pjsip: Increase maximum packet size.

The maximum packet size for PJSIP has been increased to handle the
multiple streams being added for WebRTC.

Change-Id: I9ea1e8d02668c544acadcb1c6200e1cc1bd588b3

23 months agoMerge "app_queue: Add change priority of call"
George Joseph [Tue, 18 Jul 2017 14:37:36 +0000 (09:37 -0500)]
Merge "app_queue: Add change priority of call"

23 months agoMerge "bridge_softmix: Don't reorder streams on participant leaving."
Jenkins2 [Tue, 18 Jul 2017 13:13:15 +0000 (08:13 -0500)]
Merge "bridge_softmix: Don't reorder streams on participant leaving."

23 months agoMerge "bridge/core_unreal: Fix SFU bugs with forwarding frames."
Jenkins2 [Mon, 17 Jul 2017 22:59:32 +0000 (17:59 -0500)]
Merge "bridge/core_unreal: Fix SFU bugs with forwarding frames."

23 months agoMerge "res_pjsip: Add "webrtc" configuration option"
Jenkins2 [Mon, 17 Jul 2017 20:16:30 +0000 (15:16 -0500)]
Merge "res_pjsip: Add "webrtc" configuration option"

23 months agoMerge "res_rtp_asterisk: Use RTP component for ICE if RTCP-MUX is in use."
Jenkins2 [Mon, 17 Jul 2017 19:54:22 +0000 (14:54 -0500)]
Merge "res_rtp_asterisk: Use RTP component for ICE if RTCP-MUX is in use."

23 months agobridge_softmix: Don't reorder streams on participant leaving.
Joshua Colp [Mon, 17 Jul 2017 12:19:59 +0000 (12:19 +0000)]
bridge_softmix: Don't reorder streams on participant leaving.

When a participant leaves a bridge while operating in SFU mode
their respective stream on every other participant needs to be
removed. Leaving the stream out of the new topology results in
every stream after it being moved and reordered. This causes
problems with clients. Instead simply mark the stream as removed
which leaves it in place in the SDP and doesn't reorder or touch
any other streams.

ASTERISK-27136

Change-Id: I4b3f840adcdf69b83842b0d8a737665ba0ef9cb1

23 months agoMerge "app_confbridge: Make sure name recordings are always removed from the filesystem"
Jenkins2 [Mon, 17 Jul 2017 14:35:08 +0000 (09:35 -0500)]
Merge "app_confbridge: Make sure name recordings are always removed from the filesystem"

23 months agoMerge "chan_iax2: On reload make sure to check for existing MWI subscription"
George Joseph [Mon, 17 Jul 2017 14:03:14 +0000 (09:03 -0500)]
Merge "chan_iax2: On reload make sure to check for existing MWI subscription"

23 months agoMerge "res/res_stasis_snoop: generate silence when audiohook returns null"
Jenkins2 [Mon, 17 Jul 2017 13:25:29 +0000 (08:25 -0500)]
Merge "res/res_stasis_snoop: generate silence when audiohook returns null"

23 months agobridge_softmix: Use removed stream spots when renegotiating.
Joshua Colp [Sun, 16 Jul 2017 17:31:35 +0000 (17:31 +0000)]
bridge_softmix: Use removed stream spots when renegotiating.

Streams are never truly removed in SDP, they still occupy
a location within the SDP. This location can be reused by
another stream if it so chooses.

This change takes advantage of this such that if a new stream
is needing to be added for a new participant any removed streams
are instead replaced first. This reduces the size of the SDP
and the number of streams.

ASTERISK-27134

Change-Id: I95cdcfd55cf47e02ea52abb5d94008db3fb68b1d

23 months agores_rtp_asterisk: Use RTP component for ICE if RTCP-MUX is in use.
Joshua Colp [Sun, 16 Jul 2017 17:18:39 +0000 (17:18 +0000)]
res_rtp_asterisk: Use RTP component for ICE if RTCP-MUX is in use.

This change makes it so that if an RTCP packet is being sent
the RTP ICE component is used for sending if RTCP-MUX is in use.

ASTERISK-27133

Change-Id: I6200f611ede709602ee9b89501720c29545ed68b

2 years agoMerge "Sounds: Update for core sounds 1.6 release"
Jenkins2 [Fri, 14 Jul 2017 15:56:17 +0000 (10:56 -0500)]
Merge "Sounds: Update for core sounds 1.6 release"

2 years agoapp_confbridge: Make sure name recordings are always removed from the filesystem
Sergej Kasumovic [Fri, 14 Jul 2017 06:25:36 +0000 (08:25 +0200)]
app_confbridge: Make sure name recordings are always removed from the filesystem

This commit fixes two possible scenarios:

* When recording name and if during recording you hangup, file is never
removed. This is due to the fact file location is nulled.
* When recording name and if you hangup during thank-you prompt, file
is never removed.

ASTERISK-27123 #close

Change-Id: I39b7271408b4b54ce880c5111a886aa8f28c2625

2 years agoMerge "res/res_pjsip_t38 ensure t38 requests get rejected quickly"
George Joseph [Fri, 14 Jul 2017 12:44:20 +0000 (07:44 -0500)]
Merge "res/res_pjsip_t38  ensure t38 requests get rejected quickly"

2 years agochan_iax2: On reload make sure to check for existing MWI subscription
Sergej Kasumovic [Fri, 14 Jul 2017 06:11:50 +0000 (08:11 +0200)]
chan_iax2: On reload make sure to check for existing MWI subscription

On every reload of chan_iax2 module, MWI subscription was added, which
results in additional taskprocessors being accumulated over time.

This commit fixes it by making sure we check for existing subscription
first.

This was verified with 'core show taskprocessors' CLI command.

ASTERISK-27122 #close

Change-Id: Ie2ef528fd5ca01b933eeb88188cc10967899cfb9

2 years agores_pjsip: Add "webrtc" configuration option
Kevin Harwell [Mon, 10 Jul 2017 23:17:44 +0000 (18:17 -0500)]
res_pjsip: Add "webrtc" configuration option

This patch creates a new configuration option called "webrtc". When enabled it
defaults and enables the following options that are needed in order for webrtc
to work in Asterisk:

  rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled
  media_encryption=dtls
  dtls_verify=fingerprint
  dtls_setup=actpass

When "webrtc" is enabled, this patch also parses the "msid" media level
attribute from an SDP. It will also appropriately add it onto the outgoing
session when applicable.

Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent.

ASTERISK-27119 #close

Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd

2 years agoSounds: Update for core sounds 1.6 release
Rusty Newton [Thu, 13 Jul 2017 20:43:42 +0000 (15:43 -0500)]
Sounds: Update for core sounds 1.6 release

Added necessary lines to make the en_NZ language set selectable and to get
core sounds 1.6 pulled down.

ASTERISK-26807 #close
ASTERISK-25816 #close
ASTERISK-26274 #close

Change-Id: I84e4dd4696568cc1ba318d12ac4b075461d6eed4

2 years agoMerge "res_rtp_asterisk / res_pjsip: Add support for BUNDLE."
Jenkins2 [Thu, 13 Jul 2017 19:40:11 +0000 (14:40 -0500)]
Merge "res_rtp_asterisk / res_pjsip: Add support for BUNDLE."

2 years agocore: Add PARSE_TIMELEN support to ast_parse_arg and ACO.
Corey Farrell [Mon, 10 Jul 2017 19:04:58 +0000 (15:04 -0400)]
core: Add PARSE_TIMELEN support to ast_parse_arg and ACO.

This adds support for parsing timelen values from config files.  This
includes support for all flags which apply to PARSE_INT32.  Support for
this parser is added to ACO via the OPT_TIMELEN_T option type.

Fixes an issue where extra characters provided to ast_app_parse_timelen
were ignored, they now cause an error.

Testing is included.

ASTERISK-27117 #close

Change-Id: I6b333feca7e3f83b4ef5bf2636fc0fd613742554

2 years agores_rtp_asterisk / res_pjsip: Add support for BUNDLE.
Joshua Colp [Fri, 30 Jun 2017 18:55:57 +0000 (18:55 +0000)]
res_rtp_asterisk / res_pjsip: Add support for BUNDLE.

BUNDLE is a specification used in WebRTC to allow multiple
streams to use the same underlying transport. This reduces
the number of ICE and DTLS negotiations that has to occur
to 1 normally.

This change implements this by adding support for it to
the RTP SDP module in PJSIP. BUNDLE can be turned on using
the "bundle" option and on an offer we will offer to
bundle streams together. On an answer we will accept any
bundle groups provided. Once accepted each stream is bundled
to another RTP instance for transport.

For the res_rtp_asterisk changes the ability to bundle
an RTP instance to another based on the SSRC received
from the remote side has been added. For outgoing traffic
if an RTP instance is bundled to another we will use the
other RTP instance for any transport related things. For
incoming traffic received from the transport instance we
look up the correct instance based on the SSRC and use it
for any non-transport related data.

ASTERISK-27118

Change-Id: I96c0920b9f9aca7382256484765a239017973c11

2 years agores/res_stasis_snoop: generate silence when audiohook returns null
Torrey Searle [Tue, 11 Jul 2017 14:55:36 +0000 (16:55 +0200)]
res/res_stasis_snoop: generate silence when audiohook returns null

Currently when rtp is paused, no packets are written to the
recorded audio file, causing the silence to be skipped and recording
not properly time aligned.  The read handler as been adapted to
return a silence frame of the correct size.

ASTERISK-27128 #close

Change-Id: I2d7f60650457860b9c70907b14426756b058a844

2 years agores/res_pjsip_t38 ensure t38 requests get rejected quickly
Torrey Searle [Thu, 22 Jun 2017 12:47:54 +0000 (14:47 +0200)]
res/res_pjsip_t38  ensure t38 requests get rejected quickly

arm the t38 webhook always, so we can correctly reject a
T38 negotiation request when t38 is disabled on a channel

Change-Id: Ib1ffe35aee145d4e0fe61dd012580be11aae079d

2 years agocore: Add digit filtering to ast_waitfordigit_full
Corey Farrell [Wed, 12 Jul 2017 18:24:36 +0000 (14:24 -0400)]
core: Add digit filtering to ast_waitfordigit_full

This adds a parameter to ast_waitfordigit_full which can be used to only
stop waiting when certain expected digits are received.  Any unexpected
DTMF digits are simply ignored.

This also creates a new dialplan application WaitDigit.

ASTERISK-27129 #close

Change-Id: Id233935ea3d13e71c75a0861834c5936c3700ef9

2 years agoapp_playback.c: Use the timezonename parameter
Holger Hans Peter Freyther [Tue, 11 Jul 2017 09:48:26 +0000 (11:48 +0200)]
app_playback.c: Use the timezonename parameter

In say_date_generic the timezonename parameter is passed but never
used. Fix it by passing it to the ast_localtime function.

ASTERISK-27124

Change-Id: I63106b8db10426d417d7275f22554a616e92fae4

2 years agobasic-pbx: Remove res_pjsip_multihomed from sample config
Sean Bright [Wed, 12 Jul 2017 20:07:42 +0000 (16:07 -0400)]
basic-pbx: Remove res_pjsip_multihomed from sample config

ASTERISK-27127 #close
Reported by: HZMI8gkCvPpom0tM

Change-Id: I2b0c54570d58156e37166ac536728af3b6c01789

2 years agoMerge "app_stream_echo: misc bug fixes"
Joshua Colp [Wed, 12 Jul 2017 11:13:34 +0000 (06:13 -0500)]
Merge "app_stream_echo: misc bug fixes"

2 years agoMerge "res_rtp_asterisk: trigger source change control frame when dtls is established"
Joshua Colp [Wed, 12 Jul 2017 11:13:25 +0000 (06:13 -0500)]
Merge "res_rtp_asterisk: trigger source change control frame when dtls is established"

2 years agoMerge "res_musiconhold: Add kill_escalation_delay, kill_method to class"
Joshua Colp [Wed, 12 Jul 2017 10:48:01 +0000 (05:48 -0500)]
Merge "res_musiconhold:  Add kill_escalation_delay, kill_method to class"

2 years agoMerge "manager: Remove AMI "Queues" action."
Joshua Colp [Wed, 12 Jul 2017 09:25:45 +0000 (04:25 -0500)]
Merge "manager: Remove AMI "Queues" action."

2 years agoMerge "Avoid setting maxfiles for a remote asterisk"
Joshua Colp [Wed, 12 Jul 2017 09:24:43 +0000 (04:24 -0500)]
Merge "Avoid setting maxfiles for a remote asterisk"

2 years agoMerge "http.c: Reduce log spam"
Jenkins2 [Wed, 12 Jul 2017 00:42:10 +0000 (19:42 -0500)]
Merge "http.c:  Reduce log spam"

2 years agobridge/core_unreal: Fix SFU bugs with forwarding frames.
Joshua Colp [Tue, 11 Jul 2017 19:33:44 +0000 (19:33 +0000)]
bridge/core_unreal: Fix SFU bugs with forwarding frames.

This change fixes a few things uncovered during SFU testing.

1. Unreal channels incorrectly forwarded video frames when
no video stream was present on them. This caused a crash when
they were read as the core requires a stream to exist for the
underlying media type. The Unreal channel will now ensure a
stream exists for the media type before forwarding the frame
and if no stream exists then the frame is dropped.

2. Mapping of frames during bridging from the stream number of
the underlying channel to the stream number of the bridge was
done in the wrong location. This resulted in the frame getting
dropped. This mapping now occurs on reading of the frame from
the channel.

3. Bridging was using the wrong ast_read function resulting in
it living in a non-multistream world.

4. In bridge_softmix when adding new streams to existing channels
the wrong stream topology was copied resulting in no streams
being added.

Change-Id: Ib7445722c3219951d6740802a0feddf2908c18c8

2 years agores_musiconhold: Add kill_escalation_delay, kill_method to class
George Joseph [Tue, 11 Jul 2017 12:26:27 +0000 (06:26 -0600)]
res_musiconhold:  Add kill_escalation_delay, kill_method to class

By default, when res_musiconhold reloads or unloads, it sends a HUP
signal to custom applications (and all descendants), waits 100ms,
then sends a TERM signal, waits 100ms, then finally sends a KILL
signal.  An application which is interacting with an external
device and/or spawns children of its own may not be able to exit
cleanly in the default times, expecially if sent a KILL signal, or
if it's children are getting signals directly from
res_musiconhoild.

* To allow extra time, the 'kill_escalation_delay'
  class option can be used to set the number of milliseconds
  res_musiconhold waits before escalating kill signals, with the
  default being the current 100ms.

* To control to whom the signals are sent, the "kill_method" class
  option can be set to "process_group" (the default, existing
  behavior), which sends signals to the application and its
  descendants directly, or "process" which sends signals only to the
  application itself.

Change-Id: Iff70a1a9405685a9021a68416830c0db5158603b

2 years agomanager: Remove AMI "Queues" action.
Benjamin Keith Ford [Wed, 5 Jul 2017 17:44:18 +0000 (12:44 -0500)]
manager: Remove AMI "Queues" action.

When performing the "Queues" action via AMI, it outputs the same
text that the Asterisk CLI outputs when running a "queue show"
command, which does not conform with the AMI spec. "QueueStatus"
already does what the "Queues" action should do, so instead of
correcting the output, the "Queues" action will be removed and
"QueueStatus" should be used instead.

ASTERISK-27073 #close
Reported by: Brian

Change-Id: Id11743859758255b69cc3a557750d7a56c6d16f8

2 years agoAvoid setting maxfiles for a remote asterisk
Tzafrir Cohen [Mon, 3 Jul 2017 12:30:37 +0000 (15:30 +0300)]
Avoid setting maxfiles for a remote asterisk

Setting maxfiles (maximum number of open files) has no practical
effect on a remote asterisk (rasterisk, rasterisk -x).

It has an ill effect of printing an extra message, which
may be annoying in case of -x.

ASTERISK-27105 #close

Change-Id: Iaf9eb344e4b4b517df91b736b27ec55f6a6921a2

2 years agohttp.c: Reduce log spam
George Joseph [Wed, 5 Jul 2017 20:31:43 +0000 (14:31 -0600)]
http.c:  Reduce log spam

Messages like "fwrite() failed: Connection reset by peer" are no
help whatsoever, especially since they can be caused simply by a
client disconnecting.

* Make those WARNINGs DEBUGs.
* Check the return from ast_iostream_printf of headers.

Change-Id: I17bd5f3621514152a7b2b263c801324c5e96568b

2 years agoMerge "res_pjsip: Fix crash with from_user containing invalid characters."
Jenkins2 [Tue, 11 Jul 2017 12:08:39 +0000 (07:08 -0500)]
Merge "res_pjsip: Fix crash with from_user containing invalid characters."

2 years agoMerge "json.c: Add backtrace log to find 'Invalid UTF-8 string' errors"
Jenkins2 [Mon, 10 Jul 2017 16:41:17 +0000 (11:41 -0500)]
Merge "json.c: Add backtrace log to find 'Invalid UTF-8 string' errors"

2 years agoMerge "res_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock."
Jenkins2 [Mon, 10 Jul 2017 16:19:16 +0000 (11:19 -0500)]
Merge "res_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock."

2 years agoMerge "bridge_native_rtp.c: Fix direct media video RTP instance ACL check."
Jenkins2 [Mon, 10 Jul 2017 15:53:11 +0000 (10:53 -0500)]
Merge "bridge_native_rtp.c: Fix direct media video RTP instance ACL check."

2 years agores_pjsip: Fix crash with from_user containing invalid characters.
Benjamin Keith Ford [Fri, 7 Jul 2017 16:19:13 +0000 (11:19 -0500)]
res_pjsip: Fix crash with from_user containing invalid characters.

If the from_user field contains certain characters (like @, {, ^, etc.),
PJSIP will return a null value for the URI when attempting to parse it.
This causes a crash when trying to dial out through a trunk that contains
these invalid characters in its from_user field.

This change checks the configuration and ensures that an endpoint will
not be created if the from_user contains an invalid character. It also
adds a null check to the PJSIP URI parsing as a backup.

ASTERISK-27036 #close
Reported by: Maxim Vasilev

Change-Id: I0396fdb5080604e0bdf1277464d5c8a85db913d0

2 years agoMerge "app_queue: Add priority to AMI QueueStatus"
George Joseph [Mon, 10 Jul 2017 14:50:37 +0000 (09:50 -0500)]
Merge "app_queue: Add priority to AMI QueueStatus"

2 years agojson.c: Add backtrace log to find 'Invalid UTF-8 string' errors
Richard Mudgett [Wed, 28 Jun 2017 00:27:43 +0000 (19:27 -0500)]
json.c: Add backtrace log to find 'Invalid UTF-8 string' errors

Change-Id: I9020ff9f2b3749904317c0c173f47a1bbed6f929

2 years agoMerge "app_voicemail: Cleanup ODBC connection handling"
Joshua Colp [Fri, 7 Jul 2017 21:38:21 +0000 (16:38 -0500)]
Merge "app_voicemail: Cleanup ODBC connection handling"

2 years agoMerge "core: Remove 'Data Retrieval API'"
Jenkins2 [Fri, 7 Jul 2017 20:42:56 +0000 (15:42 -0500)]
Merge "core: Remove 'Data Retrieval API'"

2 years agores_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock.
Richard Mudgett [Wed, 5 Jul 2017 18:39:45 +0000 (13:39 -0500)]
res_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock.

When a message is received on the TURN socket, the code processing the
message needs to call into the ICE/STUN session for further processing.
This code path locks the TURN group lock then the ICE/STUN group lock.  In
another thread an ICE/STUN timer can fire off to send a keep alive message
over the TURN socket.  In this code path, the ICE/STUN group lock is
obtained then the TURN group lock is obtained to send the packet.  A
classic deadlock case if the group locks are not the same.

* Made TURN get created using the ICE/STUN session's group lock.

NOTE: I was originally concerned that the ICE/STUN session can get
recreated by ice_reset_session() for an event like RTCP multiplexing
causing a change during SDP negotiation.  In this case the TURN group lock
would become different.  However, TURN is also recreated as part of the
ICE/STUN recreation in ice_create() when all known ICE candidates are
added to the new ICE session.  While the ICE/STUN and TURN sessions are
being recreated there is a period where the group locks could be
different.

ASTERISK-27023 #close
Patches:
    res_rtp_asterisk-turn-deadlock-fix.patch (license #6502)
        patch uploaded by Michael Walton (modified)

Change-Id: Ic870edb99ce4988a8c8eb6e678ca7f19da1432b9

2 years agoFix alembic branches
George Joseph [Thu, 6 Jul 2017 10:55:17 +0000 (04:55 -0600)]
Fix alembic branches

Change-Id: I04f607f084bda9b1b7f626e8e9735c37dc751187

2 years agoMerge "channel: Clear channel flag in error branch."
Joshua Colp [Wed, 5 Jul 2017 23:46:10 +0000 (18:46 -0500)]
Merge "channel: Clear channel flag in error branch."

2 years agoMerge "pjproject_bundled: Allow passing configure options to bundled"
Jenkins2 [Wed, 5 Jul 2017 22:59:39 +0000 (17:59 -0500)]
Merge "pjproject_bundled:  Allow passing configure options to bundled"

2 years agobridge_native_rtp.c: Fix direct media video RTP instance ACL check.
Richard Mudgett [Fri, 23 Jun 2017 16:17:51 +0000 (11:17 -0500)]
bridge_native_rtp.c: Fix direct media video RTP instance ACL check.

The video stream was using the audio stream RTP instance addresses to
check if the video RTP gets directed to an allowed direct media Access
Control List (ACL) address.  There is no guarantee that the video RTP
instance uses the same addresses as the audio RTP instance.

This looks like it has been a bug since v11 when direct media ACL was
first added to chan_sip and then faithfully reproduced through a couple
code refactorings into the new bridging architecture.

Change-Id: I8ddd56320e0eea769f3ceed3fa5b6bdfb51d681a

2 years agoMerge "bridge_native_rtp: Keep rtp instance refs on bridge_channel"
George Joseph [Wed, 5 Jul 2017 22:03:28 +0000 (17:03 -0500)]
Merge "bridge_native_rtp: Keep rtp instance refs on bridge_channel"

2 years agoMerge "chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support)."
Jenkins2 [Wed, 5 Jul 2017 21:37:39 +0000 (16:37 -0500)]
Merge "chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support)."

2 years agoMerge "chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support)."
Jenkins2 [Wed, 5 Jul 2017 21:29:45 +0000 (16:29 -0500)]
Merge "chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support)."

2 years agoMerge "pjsip_distributor.c: Fix deadlock with TCP type transports."
George Joseph [Wed, 5 Jul 2017 21:08:46 +0000 (16:08 -0500)]
Merge "pjsip_distributor.c: Fix deadlock with TCP type transports."

2 years agoMerge "pjsip_distributor.c: Fix unidentified_requests hash functions."
Jenkins2 [Wed, 5 Jul 2017 20:32:40 +0000 (15:32 -0500)]
Merge "pjsip_distributor.c: Fix unidentified_requests hash functions."

2 years agoMerge "chan_pjsip: Fix ability to send UPDATE on COLP"
Jenkins2 [Wed, 5 Jul 2017 19:17:23 +0000 (14:17 -0500)]
Merge "chan_pjsip:  Fix ability to send UPDATE on COLP"

2 years agocore: Remove 'Data Retrieval API'
Sean Bright [Wed, 5 Jul 2017 15:29:01 +0000 (11:29 -0400)]
core: Remove 'Data Retrieval API'

This API was not actively maintained, was not added to new modules
(such as res_pjsip), and there exist better alternatives to acquire the
same information, such as the ARI.

Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83

2 years agoapp_queue: Add change priority of call
Rodrigo Ramírez Norambuena [Mon, 19 Jun 2017 16:22:40 +0000 (12:22 -0400)]
app_queue: Add change priority of call

This patch include a feature to change the priority a caller in a
queue by CLI and AMI.

Change-Id: I55d520d71cc1cefe9a9b81fefaefc14679e96133

2 years agochan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support).
Alexander Traud [Mon, 3 Jul 2017 15:59:43 +0000 (17:59 +0200)]
chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support).

When sip.conf contained tcpenable=yes and autodomain=yes, the TCP domain was
added in any case, because of a local Boolean-negation error of the return value
of ast_sockaddr_cmp. After fixing this error for TCP and TLS, the TLS domain was
still always added with tlsenable=yes, because the domains were not compared
just on the address but also on the port – and TLS is always on a different port
than UDP/TCP.

ASTERISK-27106

Change-Id: I14fe9e319e238320b094016980445ef3a5b3337c

2 years agochan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support).
Alexander Traud [Mon, 3 Jul 2017 15:38:32 +0000 (17:38 +0200)]
chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support).

Because of a copy-and-paste error when the struct ast_sockaddr changed,
tlsbindaddr was not added, when sip.conf contained autodomain=yes; see
"show sip domains" on the command-line interface (CLI) of Asterisk.

ASTERISK-27106

Change-Id: I3d0957150017c223136968ef1266f275d0d6695e

2 years agoapp_voicemail: Cleanup ODBC connection handling
Sean Bright [Thu, 29 Jun 2017 18:58:35 +0000 (14:58 -0400)]
app_voicemail: Cleanup ODBC connection handling

The primary focus of this patch is adding a missing call to
ast_odbc_release_obj(), but is also a general cleanup of the ODBC
related code in app_voicemail.

ASTERISK-27093 #close

Change-Id: I8e285142eaeb3146b4287a928276b70db76c902b

2 years agochannel: Clear channel flag in error branch.
Corey Farrell [Sat, 1 Jul 2017 04:57:31 +0000 (00:57 -0400)]
channel: Clear channel flag in error branch.

Clear channel flag AST_FLAG_END_DTMF_ONLY in ast_waitfordigit_full when
ast_read returns NULL.

ASTERISK-27100 #close

Change-Id: Id3039e9a4e74e0cb359f636c9fd0c9740ebf7d9d